The present invention relates to improving the performance of audio equipment and in particular to adapting equalization to a speaker and room combination.
Low frequency room acoustic response modeling and equalization is a challenging problem. Traditionally, Infinite-duration Impulse Response (IIR) or Finite-duration Impulse Response (FIR) filters have been used for acoustic response modeling and equalization. The IIR filter, also called a parametric filter; has a bell-shaped magnitude response and is characterized by its center frequency Fc, the gain G at the center frequency, and a Q factor (which is inversely related to the bandwidth of the filter) and is easily implemented as a cascade for purposes of room response modeling and equalization.
Room response modeling, and hence equalization or correction, has traditionally been approached as an inverse filter problem, where the resulting equalization filter is the inverse of the room response (or the minimum phase part). Such response modeling is especially challenging at low frequencies where standing waves often cause significant variations in the frequency response at a listening position. Typical filter structures for realizable equalization filter design include IIR filters or warped FIR filters.
A typical room is an acoustic enclosure which may be modeled as a linear system. When a loudspeaker is placed in the room, the resulting time domain response is the convolution of the room linear response and the loudspeaker response, and is denoted as a loudspeaker-room impulse response h(n); nε{O, 1, 2, . . . }. The loudspeaker-room impulse response has an associated frequency response, H(ejw), which is a function of frequency. Generally, H(ejw) is also referred to as the Loudspeaker-Room Transfer Function (LRTF). In the frequency domain, the LRTF shows significant spectral peaks and dips in the human range of hearing (i.e., 20 Hz to 20 kHz) in the magnitude response, causing audible sound degradation at a listener position. FIG. 1 shows an unsmoothed LRTF plot and a ⅓-octave smoothed LRTF plot of the loudspeaker-room response. As is evident from the smoothed LRTF plot, the loudspeaker-room response exhibits a large gain of about 10 dB at 75 Hz with a peak region about an octave wide at the 3 dB down point which results in unwanted amplification of sound in the peak region. A notch region at about 145 Hz is half-octave wide and attenuates sound in the notch region. Additional variations throughout the frequency range of hearing (20 Hz-20 kHZ), and a non-smooth and non-flat envelope of the response, will result in a poor sound reproduction from the loudspeaker in the room where these measurements were made. The objective of equalization is to correct the response variations in the frequency domain (i.e., minimize the deviations in the magnitude response) and ideally also minimize the energy of the reflections in the time domain.
Known methods of equalization include modeling the room responses (either via time domain or magnitude domain or jointly) and subsequently inverting the model to obtain an equalization filter. Unfortunately, traditional search based parametric filter design approaches (such as described in “Direct Method with Random Optimization for Parametric IIR Audio Equalization” by Ramos and Lopez, Proc. 116 AES Conv., Berlin May 2004) involve a search strategy which is susceptible to being stuck in a local minima, thereby effectively limiting the amount of correction at low frequencies.